An emerging alternative to traditional telephony is Internet Protocol (IP) telephony, also known as Voice over Internet Protocol (VoIP). In contrast to traditional telephony, which uses circuit-switched protocols to establish end-to-end circuits between endpoints, IP telephony uses the Internet Protocol to achieve real-time transmission of packetized voice signals over an IP network, such as the Internet. The Internet Protocol provides connectionless, best effort delivery of datagrams or packets through the IP network. Unlike data traffic, however, real-time voice traffic is sensitive to packet loss and latency. The problem of network congestion, therefore, has different repercussions for real-time delay-sensitive traffic, such as voice, than for data traffic. Techniques such as delaying, dropping, or retransmitting data packets during network congestion may achieve satisfactory results for purposes of transmitting data, whereas such actions would be detrimental to the quality of service expected by end-users conducting a real-time IP telephone call. Consequently, it is preferable to deny access to an IP network experiencing congestion than to permit voice traffic onto the network, if the network is only going to delay or drop the voice traffic and thereby produce an unsatisfactory, if not irritating, telephone call.
To avoid such consequences, networks employ call admission control (CAC) techniques to determine, before establishing a new IP telephone call, whether network resources are available to support the call at an acceptable quality of service. In general, if the needed network resources are available, the network accepts the call; otherwise, the call is rejected from the start. In some networks, every hop in the network can participate in the decision to accept or reject the establishment of a new IP call. When individual hops in a “hop-by-hop” network individually perform CAC, the CAC mechanism can be referred to as tandem call admission control.
Implementations of IP telephony use a variety of signaling protocols, which include the H.323 family of protocols and the Session Initiation Protocol (SIP). In general, an IP telephony signaling protocol employs signaling messages to invite, accept, and reject the creation of sessions, such as Internet multimedia conferences, Internet telephone calls, and multimedia distribution. However, some IP telephony signaling protocols, (e.g., SIP), do not have tandem CAC capability, and thus cannot make hop-by-hop decisions regarding whether to accept or reject a new IP call.